The present invention relates in general to mixed analog and digital signal processing and in particular, to sample rate converters using virtual sample rates and analog to digital and digital to analog converters using the same.
In many applications, converting data from its native analog form into the digital domain for processing, storage and transmission provides the best overall system performance. One well known example is audio processing where analog audio is digitized through analog to digital (A/D) conversion and then processed, for example filtered or compressed, and then stored on a digital storage medium such as a compact disk (CD) or digital video disk (DVD). On playback, the digital data is decompressed, as required, reconverted to analog through digital to analog (D/A) conversion, and finally presented to the end user as audible tones.
Another example is digital telephony, where speech is digitally encoded at the transmitting end, carried in digital form across a digital medium such as a network, and then reconstructed at the receiving end.
According to the Nyquist Theorem, so long as the analog waveform is sampled during A/D conversion at a sampling frequency at least twice as high as the highest frequency component, that waveform can be successively reconstructed during subsequent D/A conversion. In actual practice, oversampling A/D and D/A converters are typically used because of their relative ease in implementation. For example, in an 8xc3x97 oversampling converter operating on data with a base sampling rate of 44.1 kHz, the data are sampled at a rate of 352.8 kHz. At the higher sampling rate, operations such as anti-aliasing filtering are easier since a substantial amount of the noise power is translated to frequency bands well above the band of the signal of interest.
Sample rate conversion is an additional problem which must be addressed when processing digitized analog data. Specifically, there are a number of different standard audio sampling rates, such as 48 kHz, 44.1 kHz, 22.05 kHz, 16 kHz and 8 kHz. Therefore, in order to properly interface systems operating on audio data at different ones of these rates, sample rate conversion must be performed. There are several existing sample rate conversion techniques, including decimation for lowering the sampling rate and interpolation for increasing the sampling rate. Notwithstanding, these techniques are still subject to some significant disadvantages including the need for substantial silicon area for fabricating the requisite interpolation/decimation filters, as well as limitations on the ability to convert to fractional sampling rates.
The principles of the present invention are embodied in sample rate conversion circuits and methods. According to one such method, sample rate conversion is performed in a data converter operating from an oversampling. clock corresponding to a native sample rate and a native oversampling factor. A virtual sample rate and a virtual oversampling factor are selected to be proportional to the native sample rate and the native oversampling factor. A data stream having a data sample rate is sampled by the virtual oversampling factor. The data stream is also resampled with a resampling ratio approximating a ratio of the data sample rate to the virtual sample rate.
Circuits and methods embodying the inventive principles allow data streams of different sample.rates to be processed in systems clocked at clock frequencies which do not necessarily match one or more of these sample rates. For example, methods are disclosed for processing standard sample rate audio data in an audio codec operating from an oversampling clock which is a sub-multiple of the standard Universal Serial. Bus clock. This oversampling clock when used to oversample at a standard oversampling ratio results in some instances in a native sample rate is less than the sample rate of the audio data being converted. The present methods and circuits address the problem of this mismatch.